Delta-Sigma Modulators │ ├─ Analog ΔΣ Modulators │ │ │ ├─ Discrete-Time Analog ΔΣ Modulator, DT ΔΣM │ │ ├─ Switched-capacitor / switched-current loop filter │ │ ├─ Input is sampled before or inside loop filter │ │ └─ Very common in ΔΣ ADCs │ │ │ └─ Continuous-Time Analog ΔΣ Modulator, CT ΔΣM │ ├─ Continuous-time RC / gm-C / active-RC loop filter │ ├─ Feedback DAC is continuous-time │ ├─ Quantizer is usually clocked → synchronous CT ΔΣM │ └─ Quantizer/comparator may be event-driven → asynchronous CT ΔΣM │ └─ Digital ΔΣ Modulators, DDSM │ ├─ Single-quantizer DDSM │ ├─ Output-feedback form │ └─ Error-feedback form, EFM │ └─ MASH DDSM └─ Multi-stage noise-shaping modulator
"Quantizers" and "truncators",
"integrators" and "accumulators" are
used in delta-sigma ADCs and DACs,
respectively
P. Kiss, J. Arias and Dandan Li, "Stable high-order delta-sigma
DACS," 2003 IEEE International Symposium on Circuits and Systems
(ISCAS), Bangkok, 2003 [https://www.ele.uva.es/~jesus/analog/tcasi2003.pdf]
plot(w1/2/pi, abs(h1), LineWidth=3) hold on plot(w2/2/pi, abs(h2), LineWidth=3) grid on legend('MOD1', 'MOD2') xlabel('fs') ylabel('mag') title('NTF of MOD1 & MOD2')
SQNR improvement
In general, for an \(l\)th order
modulator with \(\text{NTF}(z) = (1 −
z^{−1})^l\), the SQNR increases by \((6l + 3)\) dB for every
doubling of the OSR, which provides \(l+0.5\)extra bits
resolution
\(\Delta \Sigma\) modulators,
and other noise-shaping modulators, change the spectrum of the
noise but leave the signal unchanged
\(\Delta\) modulators and other
signal-predicting modulators shape the spectrum of the
modulated signal but leave the quantization noise unchanged at the
receiver
output vs.
error-feedback
The error-feedback architecture is
problematic for analog implementation, since it is
sensitive to variations of its parameters (subtractor
realization)
ADC
DAC
P. Kiss, J. Arias and Dandan Li, "Stable high-order delta-sigma
DACS," 2003 IEEE International Symposium on Circuits and Systems
(ISCAS), Bangkok, 2003 [https://www.ele.uva.es/~jesus/analog/tcasi2003.pdf]
always @(*) i_func_extended = {i_func[15],i_func[15],i_func[15],i_func[15],i_func}; always @(posedge i_clk ornegedge i_res) begin if (i_res==0) begin DAC_acc_1st<=16'd0; DAC_acc_2nd<=16'd0; this_bit = 1'b0; end elseif(i_ce == 1'b1) begin if(this_bit == 1'b1) begin DAC_acc_1st = DAC_acc_1st + i_func_extended - (2**15); DAC_acc_2nd = DAC_acc_2nd + DAC_acc_1st - (2**15); end else begin DAC_acc_1st = DAC_acc_1st + i_func_extended + (2**15); DAC_acc_2nd = DAC_acc_2nd + DAC_acc_1st + (2**15); end // When the high bit is set (a negative value) we need to output a 0 and when it is clear we need to output a 1. this_bit = ~DAC_acc_2nd[19]; end end endmodule
an interpolation filter effectively
up-samples its low-rate input and
lowpass-filters the resulting high-rate data
to produce a high-rate output devoid of images
Any such physically feasible device will take
a finite time to operate – in other words, the
quantized output will only be available a small time
after the quantizer has "looked" at the input - insert a one-sample
delay
there cannot be a "delay free loop" is a
common idea in sequential digital state machine design
Both integrator and quantizer are delay free
NTF realizability criterion: No delay-free loops in the modulator
linear settling & GBW of
amplifier
TODO 📅
Switched capacitor has been the common realization technique of
discrete-time (DT) modulators, and in order to achieve a
linear settling, the sampling frequency used
in these converters needs to be significantly lower than the gain
bandwidth product (GBW) of the amplifiers.
function[v,xn,xmax,y] = simulateDSM(u,arg2,nlev,x0) %[v,xn,xmax,y] = simulateDSM(u,ABCD,nlev=2,x0=0) % or %[v,xn,xmax,y] = simulateDSM(u,ntf,nlev=2,x0=0) % %Compute the output of a general delta-sigma modulator with input u, %a structure described by ABCD, an initial state x0 (default zero) and %a quantizer with a number of levels specified by nlev. %Multiple quantizers are implied by making nlev an array, %and multiple inputs are implied by the number of rows in u. % %Alternatively, the modulator may be described by an NTF. %The NTF is zpk object. (The STF is assumed to be 1.) %The structure that is simulated is the block-diagional structure used by %zp2ss.m.
calculateSNR
1 2 3 4 5 6 7 8 9 10 11 12
functionsnr = calculateSNR(hwfft,f,nsig)
signalBins = [f-nsig+1:f+nsig+1]; s = norm(hwfft(signalBins));
noiseBins = 1:length(hwfft); noiseBins(signalBins) = []; n = norm(hwfft(noiseBins));
snr = dbv(s/n);
end
reference
Pavan, Shanthi, Richard Schreier, and Gabor Temes. (2016).
Understanding Delta-Sigma Data Converters. 2nd ed. Wiley.
Norsworthy, Steven R., Richard Schreier, Gábor C. Temes and Ieee
Circuits. “Delta-sigma data converters : theory, design, and
simulation.” (1997).
Horowitz, P., & Hill, W. (2015). The art of electronics
(3rd ed.). Cambridge University Press.
John Rogers, Calvin Plett, and Foster Dai. 2006. Integrated Circuit
Design for High-Speed Frequency Synthesis (Artech House Microwave
Library). Artech House, Inc., USA.
Razavi B. Analysis and Design of Data Converters. Cambridge
University Press; 2025.
R. Schreier, ISSCC2006 tutorial: Understanding Delta-Sigma Data
Converters
P. M. Aziz, H. V. Sorensen and J. vn der Spiegel, "An overview of
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V. Medina, P. Rombouts and L. Hernandez-Corporales, "A Different View
of Sigma-Delta Modulators Under the Lens of Pulse Frequency Modulation
[Feature]," in IEEE Circuits and Systems Magazine, vol. 24, no.
2, pp. 80-97, Secondquarter 2024
Noise Fmax sets the bandwidth of the random noise
sources that are injected at each time point in the transient
analysis
We can identify the RMS noise value easily by looking at 15.9% or
84.1% of CDF (\(1\sigma\)), the
input-referred noise in the RMS is 0.9mV
Thus, if \(V_S\) is chosen so as to
reduce the probability of zeros to 16%, then \(V_S = 1\sigma\), which is also the total
root-mean square (rms) noise referred to the input.
Comparison of two methods
here, fundamental frequency = fclk; integrated noise (0 ~
0.5fclk)
E. Gillen, G. Panchanan, B. Lawton and D. O'Hare, "Comparison of
transient and PNOISE simulation techniques for the design of a dynamic
comparator," 2022 33rd Irish Signals and Systems Conference (ISSC),
Cork, Ireland, 2022, pp. 1-5
J. Conrad, J. Kauffman, S. Wilhelmstatter, R. Asthana, V. Belagiannis
and M. Ortmanns, "Confidence Estimation and Boosting for
Dynamic-Comparator Transient-Noise Analysis," 2024 22nd IEEE
Interregional NEWCAS Conference (NEWCAS), Sherbrooke, QC, Canada,
2024, pp. 1-5
There are some ambiguity in formula in ADC Verification Rapid
Adoption Kit (RAK)(Product Version: IC 6.1.8, SPECTRE 18.1 March,
2019)
Transient Noise Analysis: \(\sqrt{2}\sigma\), why ratio \(\sqrt{2}\) ???
L. Kull et al., "A 3.1 mW 8b 1.2 GS/s Single-Channel
Asynchronous SAR ADC With Alternate Comparators for Enhanced Speed in 32
nm Digital SOI CMOS," in IEEE Journal of Solid-State Circuits,
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P. Nuzzo, F. De Bernardinis, P. Terreni and G. Van der Plas, "Noise
Analysis of Regenerative Comparators for Reconfigurable ADC
Architectures," in IEEE Transactions on Circuits and Systems I:
Regular Papers, vol. 55, no. 6, pp. 1441-1454, July 2008 [https://sci-hub.jp/10.1109/TCSI.2008.917991]
If the input referred offset follows a normal distribution than it is sufficient to apply a single offset voltage to calculate the offset voltage. See details in Razavi, B., The StrongARM Latch [A Circuit for All Seasons], IEEE Solid-State Circuits Magazine, Volume:7, Issue: 2, Spring 2015
Omran, Hesham. (2019). Fast and accurate technique for comparator
offset voltage simulation. Microelectronics Journal. 89.
10.1016/j.mejo.2019.05.004.
Kickback noise trades with the dimensions of the input
transistors and hence with the offset voltage
affects the comparator's own decision
corrupts the input voltage while it is sensed by other circuits
Tetsuya Iizuka,VLSI2021_Workshop3 "Nyquist A/D Converter Design in
Four Days"
Figueiredo, Pedro & Vital, João. (2006). Kickback noise reduction
techniques for CMOS latched comparators. Circuits and Systems II:
Express Briefs, IEEE Transactions on. 53. 541 - 545.
10.1109/TCSII.2006.875308. [https://sci-hub.se/10.1109/TCSII.2006.875308]
P. M. Figueiredo and J. C. Vital, "Low kickback noise techniques for
CMOS latched comparators," 2004 IEEE International Symposium on Circuits
and Systems (ISCAS), Vancouver, BC, Canada, 2004, pp. I-537 [https://sci-hub.se/10.1109/ISCAS.2004.1328250]
Current mirrors are used between stages to reduce
charge kick back from the logic level swing of the
latch onto the small comparator input capacitors
Mike Shuo-Wei Chen and R. W. Brodersen, "A 6-bit 600-MS/s 5.3-mW
Asynchronous ADC in 0.13-μm CMOS," in IEEE Journal of Solid-State
Circuits, vol. 41, no. 12, pp. 2669-2680, Dec. 2006 [pdf,
slides]
K. Bult and A. Buchwald, "An embedded 240-mW 10-b 50-MS/s CMOS ADC in
1-mm/sup 2/," in IEEE Journal of Solid-State Circuits, vol. 32, no. 12,
pp. 1887-1895, Dec. 1997 [https://sci-hub.st/10.1109/4.643647]
If the comparator can not generate a well-defined logical output in
half of the clock period, we say the circuit is
"metastable"
Mathematical Preliminaries
Relating \(\Phi\) and erf
Error Function (Erf) of the
standard Normal distribution \[
\text{Erf}(x) = \frac{2}{\sqrt{\pi}}\int_0^x e^{-t^2} \mathrm{d}t.
\]Cumulative Distribution Function
(CDF) of the standard Normal distribution \[
\Phi(x) = \frac{1}{\sqrt{2\pi}}\int_{-\infty}^x e^{-z^2/2} \mathrm{d}z.
\]
P. Nuzzo, F. De Bernardinis, P. Terreni and G. Van der Plas, "Noise
Analysis of Regenerative Comparators for Reconfigurable ADC
Architectures," in IEEE Transactions on Circuits and Systems I:
Regular Papers, vol. 55, no. 6, pp. 1441-1454, July 2008 [https://picture.iczhiku.com/resource/eetop/SYirpPPPaAQzsNXn.pdf]
J. Kim, B. S. Leibowitz and M. Jeeradit, "Impulse sensitivity
function analysis of periodic circuits," 2008 IEEE/ACM International
Conference on Computer-Aided Design, 2008, pp. 386-391, doi:
10.1109/ICCAD.2008.4681602. [https://websrv.cecs.uci.edu/~papers/iccad08/PDFs/Papers/05C.2.pdf]
Y. Luo, A. Jain, J. Wagner and M. Ortmanns, "Input Referred
Comparator Noise in SAR ADCs," in IEEE Transactions on Circuits and
Systems II: Express Briefs, vol. 66, no. 5, pp. 718-722, May 2019. [https://sci-hub.se/10.1109/TCSII.2019.2909429]
X. Tang et al., "An Energy-Efficient Comparator With Dynamic Floating
Inverter Amplifier," in IEEE Journal of Solid-State Circuits, vol. 55,
no. 4, pp. 1011-1022, April 2020 [https://sci-hub.se/10.1109/JSSC.2019.2960485]
C. Mangelsdorf, "Metastability: Deeply misunderstood [Shop Talk: What
You Didn’t Learn in School]," in IEEE Solid-State Circuits Magazine,
vol. 16, no. 2, pp. 8-15, Spring 2024
Rabuske, Taimur & Fernandes, Jorge. (2014). Noise-aware
simulation-based sizing and optimization of clocked comparators. Analog
Integr. Circuits Signal Process.. 81. 723-728.
10.1007/s10470-014-0428-4. [https://sci-hub.se/10.1007/s10470-014-0428-4]
Rabuske, Taimur & Fernandes, Jorge. (2016). Charge-Sharing SAR
ADCs for Low-Voltage Low-Power Applications.
10.1007/978-3-319-39624-8.
Masaya Miyahara, Yusuke Asada, Daehwa Paik and Akira Matsuzawa, "A
low-noise self-calibrating dynamic comparator for high-speed ADCs,"
2008 IEEE Asian Solid-State Circuits Conference, Fukuoka,
Japan, 2008 [slides,
paper]
Igor Freire. Symbol Timing Synchronization: A Tutorial [blog,
code]
But the problem here is: "How does the receiver know the ideal
sampling instants?". The solution is "someone has to supply those
ideal sampling instants". A symbol time
recovery circuit is used for this purpose.
Early/Late Symbol Recovery algorithm
non-decision-directed timing estimator exploits the
symmetry properties of the signal
If the Early Sample = Late Sample : The
peak occurs at the on-time sampling instant \(T\). No adjustment in the timing is
needed.
If |Early Sample| > |Late Sample| :
Late timing, the sampling time is offset so that the next symbol is
sampled \(T-\delta/2\) seconds after
the current sampling time.
If |Early Sample| < |Late Sample| :
Early timing, the sampling time is offset so that the next symbol is
sampled \(T+\delta/2\) seconds after
the current sampling time.
David Johns. ECE1392H - Integrated Circuits for Digital
Communications - Fall 2001: [Timing
Recovery]
Dither in Quantized Zero Crossing Detection (QZCD)
(so-called 'Bang Bang' Phase Detector)
Mueller and Muller
Timing Synchronization
K. Mueller and M. Muller, "Timing Recovery in Digital Synchronous
Data Receivers," in IEEE Transactions on Communications, vol.
24, no. 5, pp. 516-531, May 1976 [pdf]
C.-P. Tzeng, D. Hodges and D. Messerschmitt, "Timing Recovery in
Digital Subscriber Loops Using Baud-Rate Sampling," in IEEE Journal
on Selected Areas in Communications, vol. 4, no. 8, pp. 1302-1311,
November 1986 [pdf]
H. Meyr, M. Moeneclaey, and S. A. Fechtel. "Digital Communication
Receivers: Synchronization, Channel Estimation, and Signal Processing."
Wiley [pdf]
T. Musah and A. Namachivayam, "Robust Timing Error Detection for
Multilevel Baud-Rate CDR," in IEEE Transactions on Circuits and Systems
I: Regular Papers, vol. 69, no. 10, pp. 3927-3939, Oct. 2022 [https://sci-hub.jp/10.1109/TCSI.2022.3191740]
Fulvio Spagna, CICC2018 Clock and Data Recovery Systems [pdf]
TODO 📅
Intersymbol Interference
(ISI)
L.W. Couch, Digital and Analog CommunicationSystems, 8th Edition, Pearson, 2013. [pdf]
Nyquist discovered three different methods for pulse shaping that
could be used to eliminate ISI
Nyquist's First Method (Zero ISI): physically
unrealizable (i.e., the impulse response would be noncausal and of
infinite duration), inaccurate sync will cause ISI
Nyquist's second method: allows some ISI to be
introduced in a controlled way
Nyquist's third method: area under the \(h_e(t)\) pulse within the desired symbol
interval, \(T_s\), is not zero, but the
areas under \(h_e(t)\) in adjacent
symbol intervals are zero
Nyquist Criterion & Pulses
David A. Johns, ECE1392H - Integrated Circuits for Digital
Communications - Fall 2001 [System
Overview]
Matched-Filter (MF)
David A. Johns, ECE1392H - Integrated Circuits for Digital
Communications - Fall 2001 [System
Overview]
David A. Johns, ECE1392H - Integrated Circuits for Digital
Communications - Fall 2001 [Introduction]
LMS & its
Quantized-Error Algorithms
Bruno Lima, Adaptive filtering in Python Implementations based on
Adaptive Filtering: Algorithms and Practical Implementation (Paulo S. R.
Diniz). [https://github.com/BruninLima/PydaptiveFiltering]
Sen M. Kuo. Real-Time Digital Signal Processing: Fundamentals,
Implementations and Applications, 3rd Edition. John Wiley & Sons
2013
Stankovic, Ljubisa. (2015). Digital Signal Processing with Selected
Topics.
Paulo S. R. Diniz, Adaptive Filtering: Algorithms and Practical
Implementation, 5th edition [pdf],
[matlab],
[python]
B. Farhang-Boroujeny (2013), Adaptive Filters: Theory and
Applications (2nd ed.). John Wiley & Sons, Inc.
Simon O. Haykin (2014), "Adaptive Filter Theory" Prentice-Hall, Inc.
5rd edition
A. Chan Carusone and D. A. Johns, "Analog Filter Adaptation Using a
Dithered Linear Search Algorithm," IEEE Int. Symp. Circuits and
Syst., May 2002. [PDF], [Slides]
Let the periodic waveform \(w(t)\),
with period \(T_0=2\pi/\omega_0\), have
the complex Fourier series \[
w(t)=\sum_{\ell=-\infty}^{\infty}W[\ell]e^{j\ell\omega_0t}.
\] We want the Fourier coefficients of \[
w(t)\sin(\omega_0t)
\quad\text{and}\quad
w(t)\cos(\omega_0t).
\]
Multiplication by \(\sin(\omega_0t)\)
Use \[
\sin(\omega_0t)
=
\frac{e^{j\omega_0t}-e^{-j\omega_0t}}{2j}.
\] Then \[
\begin{aligned}
w(t)\sin(\omega_0t)
&=
\left(\sum_{\ell=-\infty}^{\infty}
W[\ell]e^{j\ell\omega_0t}\right)
\frac{e^{j\omega_0t}-e^{-j\omega_0t}}{2j}
\\[4pt]
&=
\frac{1}{2j}
\sum_{\ell=-\infty}^{\infty}
W[\ell]
\left[
e^{j(\ell+1)\omega_0t}
-
e^{j(\ell-1)\omega_0t}
\right].
\end{aligned}
\] Now collect the coefficient multiplying \(e^{jk\omega_0t}\).
For the first term, \[
k=\ell+1
\quad\Longrightarrow\quad
\ell=k-1,
\] so its coefficient is \(W[k-1]\).
For the second term, \[
k=\ell-1
\quad\Longrightarrow\quad
\ell=k+1,
\] so its coefficient is \(W[k+1]\).
Therefore, \[
\boxed{
w(t)\sin(\omega_0t)
=
\sum_{k=-\infty}^{\infty}
\frac{W[k-1]-W[k+1]}{2j}
e^{jk\omega_0t}
}
\] and the Fourier coefficient is \[
\color{blue}\boxed{
\left[w(t)\sin(\omega_0t)\right]_k
=
\frac{W[k-1]-W[k+1]}{2j}.
}
\]
DFT \(X[k]\) is a
sampled version of the DTFT \(X(e^{j\hat{\omega}})\), with the \(k\)-th sampled digital frequency, \(\hat{\omega}_k = \frac{2\pi k}{N}\)\[
\boxed{X[k] = X(e^{j\hat{\omega}})\bigg|_{\hat{\omega}=\hat{\omega}_k},
\qquad
\hat{\omega}_k = \frac{2\pi k}{N},
\qquad
k=0,1,\dots,N-1}
\]
DTFT vs DFT
Schuller, Gerald. 2026. Multirate Signal Processing with Examples in
Python. Cham: Springer Nature Switzerland.
impulse train
CTFT:
using time-sampling property
DTFT:
Given \(x[n]=\sum_{k=-\infty}^{\infty}\delta(n-k)\)
where \(H(j\omega)\), \(H(e^{j\hat{\omega}})\) is frequency
response of continuous-time systems and
discrete-time systems, which is the function of \(\omega\) and \(\hat{\omega}\)\[\begin{align}
H(j\omega) &= \int_{-\infty}^{+\infty}h(t)e^{-j\omega t}dt \\ \\
H(e^{j\hat{\omega}}) &=
\sum_{n=-\infty}^{+\infty}h[n]e^{-j\hat{\omega} n}
\end{align}\]
The frequency response of discrete-time LTI systems
is always a periodic function of the frequency variable
\(\hat{\omega}\) with period \(2\pi\)
Sampling Theorem
time-sampling theorem: applies to bandlimited
signals
spectral sampling theorem: applies to
timelimited signals
Aliasing
Given below sequence \[
X[n] =A e^{j\omega T_s n}
\]
whileTrue: for signN, signFsig in product([-1, 1], [-1, 1]): fdisp_n = signN*N*fs + signFsig*fsig if fdisp_n >= 0and fdisp_n < fs/2: fdisp = fdisp_n print(f"{fsig:.4f} is indistinguishable from {fdisp:.4f}, which is {'+'if signN>0else'-'}{N}{'+'if signFsig>0else'-'}{fsig:.4f}") return fdisp N += 1 if N > 100: break returnNone
for i inrange(1,6): samplealiasing(0.3125*i)
# 0.3125 is indistinguishable from 0.3125, which is -0 + 0.3125 # 0.6250 is indistinguishable from 0.3750, which is +1 - 0.6250 # 0.9375 is indistinguishable from 0.0625, which is +1 - 0.9375 # 1.2500 is indistinguishable from 0.2500, which is -1 + 1.2500 # 1.5625 is indistinguishable from 0.4375, which is +2 - 1.5625
# inspired by https://github.com/bmurmann/MEAD2026/blob/main/tb_boot_bottom_4.ipynb
defsamplealiasing(fsig, fs=1): fdisp = None fsig_wrap = fsig % fs if fsig_wrap > fs/2: fdisp = fs - fsig_wrap else: fdisp = fsig_wrap print(f"{fsig:.4f} is indistinguishable from {fdisp:.4f}") return fdisp
for i inrange(1,6): samplealiasing(0.3125*i)
# 0.3125 is indistinguishable from 0.3125 # 0.6250 is indistinguishable from 0.3750 # 0.9375 is indistinguishable from 0.0625 # 1.2500 is indistinguishable from 0.2500 # 1.5625 is indistinguishable from 0.4375
CTFS & CTFT
Fourier transform of a periodic signal with Fourier series
coefficients \(\{a_k\}\) can be
interpreted as a train of impulses occurring at the
harmonically related frequencies and for which the area of the impulse
at the \(k\)th harmonic frequency \(k\omega_0\) is \(2\pi\) times the \(k\)th Fourier series coefficient \(a_k\)
inverse CTFT & inverse DTFT
time domain
frequency domain
inverse CTFT
\(\delta(t)\)
\(\int_{\infty}d\omega\)
inverse DTFT
\(\delta[n]\)
\(\int_{2\pi}d\hat{\omega}\)
inverse CTFT shall integral from \(-\infty\) to \(+\infty\) to obtain \(\delta(t)\) in time domain, e.g., \(x_s(t)\) impulse train
Consider the periodic pulse function \(x_T(t) = \Pi_T \left( \frac{t}{T_p}
\right)\)
Fourier Series Coefficients is \[
c_n = \frac{T_p}{T} \operatorname{sinc} \left( \frac{n T_p}{T} \right)
\] The Fourier Transform of the function is \[
X_T(\omega) = \sum_{n=-\infty}^{+\infty} c_n 2\pi \delta(\omega -
n\omega_0)
\]
spectral sampling
spectral sampling by \(\omega_0\),
and \(\frac{2\pi}{\omega_0} \gt \tau\)\[
X_{n\omega_0}(\omega) =
\sum_{n=-\infty}^{\infty}X(n\omega_0)\delta(\omega - n\omega_0)
\] Periodic repetition of \(x(t)\) is \[
x_{n\omega_0}(t) = \frac{1}{\omega_0}\sum_{n=-\infty}^{\infty}x(t
-n\frac{2\pi}{\omega_0})=\frac{T_0}{2\pi}\sum_{n=-\infty}^{\infty}x(t
-nT_0)
\]
Then, if \(x_{T_0} (t)\), a periodic
signal formed by repeating \(x(t)\)
every \(T_0\) seconds (\(T_0 \gt \tau\)), its CTFT is \[
X_{T_0}(\omega) = \frac{2\pi}{T_0} \cdot X_{n\omega_0}(\omega) =
\frac{2\pi}{T_0}\sum_{n=-\infty}^{\infty}X(n\omega_0)\delta(\omega -
n\omega_0)
\] Then \(x_{T_0} (t)\) can be
expressed with inverse CTFT as \[\begin{align}
x_{T_0} (t) &=
\frac{1}{2\pi}\int_{-\infty}^{\infty}X_{T_0}(\omega)e^{j\omega t}d\omega
\\
&= \frac{1}{T_0}\sum_{n=-\infty}^{\infty}X(n\omega_0)e^{jn\omega_0
t} =\sum_{n=-\infty}^{\infty}\frac{1}{T_0}X(n\omega_0)e^{jn\omega_0 t}
\end{align}\]
i.e. the coefficients of the Fourier series for \(x_{T_0} (t)\) is \(D_n =\frac{1}{T_0}X(n\omega_0)\)
alternative method by direct Fourier series
Why DFT ?
We can use DFT to compute DTFT samples and CTFT samples
\[
\overline{x}(t) = \sum_{n=0}^{N_0-1}x(nT)\delta(t-nT)
\] applying the Fourier transform yieds \[
\overline{X}(\omega) = \sum_{n=0}^{N_0-1}x[n]e^{-jn\omega T}
\] But \(\overline{X}(\omega)\),
the Fourier transform of \(\overline{x}(t)\) is \(X(\omega)/T\), assuming negligible
aliasing. Hence, \[
X(\omega) = T\overline{X}(\omega) = T\sum_{n=0}^{N_0-1}x[n]e^{-jn\omega
T}
\] and \[
X(k\omega_0) = T\sum_{n=0}^{N_0-1}x[n]e^{-jn k\omega_0 T}
\] with \(\hat{\omega}_0 = \omega_0
T\)\[
X(k\omega_0) = T\sum_{n=0}^{N_0-1}x[n]e^{-jn k\hat{\omega}_0}
\]i.e. the relationship between CTFT and DFT is \(X(k\omega_0) = T\cdot X[k]\), DFT is a tool
for computing the samples of CTFT
C/D
Sampling with a periodic impulse train, followed by conversion to a
discrete-time sequence
The periodic impulse train is \[
s(t) = \sum_{n=-\infty}^{\infty}\delta(t-nT)
\]\(x_s(t)\) can be expressed
as \[
x_s(t) = \sum_{n=-\infty}^{\infty}x_c(nT)\delta(t-nT)
\] i.e., the size (area) of the impulse at sample time
\(nT\) is equal to the value of the
continuous-time signal at that time.
\(x_s(t)\) is, in a sense, a
continuous-time signal (specifically, an impulse train)
samples of \(x_c(t)\) are represented by
finite numbers in \(x[n]\)
rather than as the areas of impulses, as with \(x_s(t)\)
Frequency-Domain
Representation of Sampling
The relationship between the Fourier transforms of the input and the
output of the impulse train modulator \[
X_s(j\omega) = \frac{1}{T}\sum_{k=-\infty}^{\infty}X_c(j(\omega
-k\omega_s))
\] where \(\omega_s\) is the
sampling frequency in radians/s
\(X(e^{j\hat{\omega}})\), the
discrete-time Fourier transform (DTFT) of the sequence \(x[n]\), in terms of \(X_s(j\omega)\) and \(X_c(j\omega)\)
Zero padding improves frequency grid resolution, not
spectral resolution
A smoother spectrum is not more information — it is better
interpolation of the same information.
To truly improve spectral resolution, you must observe the signal
longer (increase N).
Gotcha
A remarkable fact of linear systems is that the complex
exponentials are eigenfunctions of a linear
system, as the system output to these inputs equals the input multiplied
by a constant factor.
Both amplitude and phase may change
but the frequency does not change
For an input \(x(t)\), we can
determine the output through the use of the convolution integral, so
that with \(x(t) = e^{st}\)\[\begin{align}
y(t) &= \int_{-\infty}^{+\infty}h(\tau)x(t-\tau)d\tau \\
&= \int_{-\infty}^{+\infty} h(\tau) e^{s(t-\tau)}d\tau \\
&= e^{st}\int_{-\infty}^{+\infty} h(\tau) e^{-s\tau}d\tau \\
&= e^{st}H(s)
\end{align}\]
Take the input signal to be a complex exponential of the form \(x(t)=Ae^{j\phi}e^{j\omega t}\)
The real cosine signal is actually composed of two
complex exponential signals: one with positive
frequency and the other with negative \[
cos(\omega t + \phi) = \frac{e^{j(\omega t + \phi)} + e^{-j(\omega t +
\phi)}}{2}
\]
The sinusoidal response is the sum of the complex-exponential
response at the positive frequency \(\omega\) and the response at the
corresponding negative frequency \(-\omega\) because of LTI systems's
superposition property
input: \[\begin{align}
x(t) &= A cos(\omega t + \phi) \\
&= \frac{1}{2}Ae^{\phi}e^{\omega t} +
\frac{1}{2}Ae^{-\phi}e^{-\omega t}
\end{align}\]
J. Zhong, Y. Zhu, S. -W. Sin, S. -P. U and R. P. Martins, "Thermal
and Reference Noise Analysis of Time-Interleaving SAR and
Partial-Interleaving Pipelined-SAR ADCs," in IEEE Transactions on
Circuits and Systems I: Regular Papers, vol. 62, no. 9, pp. 2196-2206,
Sept. 2015 [https://sci-hub.st/10.1109/TCSI.2015.2452331]
The amplitude of the reference ripple is code-dependent as it is
correlated with switching energy in each bit cycling
SAR ADC Noise Analysis
kT/C Noise in sampling
DAC Noise in conversion
T. Miki et al., "A 4.2 mW 50 MS/s 13 bit CMOS SAR ADC With SNR and
SFDR Enhancement Techniques," in IEEE Journal of Solid-State Circuits,
vol. 50, no. 6, pp. 1372-1381, June 2015 [https://sci-hub.jp/10.1109/JSSC.2015.2417803]
Comparator Noise in
conversion
noise analysis for dynamic integrator
noise analysis for latch phase
P. Nuzzo, F. De Bernardinis, P. Terreni and G. Van der Plas, "Noise
Analysis of Regenerative Comparators for Reconfigurable ADC
Architectures," in IEEE Transactions on Circuits and Systems I:
Regular Papers, vol. 55, no. 6, pp. 1441-1454, July 2008
Comparator
Comparator input cap effect
\[
-V_{in}\cdot 2^N C = V_c (2^N C + C_p)
\] Then \(V_c = -\frac{2^N C}{2^N C +
C_p}V_{in}\), i.e. this capacitance reduce the voltage amplitude
by the factor
During conversion \[\begin{align}
V_c &= -\frac{2^N C}{2^N C + C_p}V_{in} +V_{ref}\sum_{n=0}^{N-1}
\frac{b_n\cdot2^n C}{2^N C + C_p} \\
&= \frac{2^N C}{2^N C + C_p}\left(-V_{in} +
V_{ref}\sum_{n=0}^{N-1}\frac{b_n }{2^{N-n}} \right)
\end{align}\]
That is, it does not change the sign
Comparator offset effect
Capacitor DAC (CDAC)
The charge redistribution capacitor network is used to
sample the input signal and serves as a digital-to-analog converter
(DAC) for creating and subtracting reference voltages
inverse Laplace Transform is \(V_y(t) =
\frac{C_1}{C_1+C_2}\left(1 - e^{-t/\tau}\right)\)
\(V_x(t)\) and \(V_y(t)\) prove that the settling time is
same
\(\tau = R\frac{C_1C_2}{C_1+C_2}\),
which means usually worst for MSB capacitor (largest)
both \(\tau\) and \(\Delta V\) are the maximum
A popular way to improve the settling behavior, again, is to employ
unit-element DACs that statistically reduce the switching activities,
which, unfortunately, exhibits unnecessary complications to the power,
area and speed tradeoffs of the design
In a SAR conversion the DAC doesn't move by full
scale — the MSB trial is the largest single
step, and it is exactly half of full scale.
Subsequent trials step by \(V_{FS}/4\),
\(V_{FS}/8\), … So the MSB transition
(\(\color{red}0 \to V_{FS}/2\)) is the
worst case, and if it settles in the allotted per-bit time, every later
trial does too.
The accuracy
criterion
The settling error must stay below half an LSB, where \(\text{LSB} = V_{FS}/2^{n}\):
Rabuske, Taimur & Fernandes, Jorge. (2016). "Appendix:
Voltage and Energy in CR ADCs", Charge-Sharing SAR ADCs for
Low-Voltage Low-Power Applications. 10.1007/978-3-319-39624-8.
That make sense, charge redistribution consume
energy
Binary-Weighted (BW) DAC
During \(\Phi_1\), all capacitor are
shorted, the net charge at \(V_x\) is
0
During \(\Phi_2\), the charge at
bottom plate of CDAC \[
Q_{DAC,btm} = \sum_{i=0}^{N-1}(b_i\cdot V_R - V_x)\cdot 2^{i}C_u =
C_uV_R\sum_{i=0}^{N-1}b_i2^i - (2^N-1)C_uV_x
\] the charge at the internal plate of integrator \[
Q_{intg} = V_x C_p + (V_x - V_o)2^NC_u
\] and we know \(-V_x A = V_o\)
and \(Q_{DAC,btm} = Q_{intg}\)\[
C_uV_R\sum_{i=0}^{N-1}b_i2^i - (2^N-1)C_uV_x = V_x C_p + (V_x -
V_o)2^NC_u
\] i.e. \[
C_uV_R\sum_{i=0}^{N-1}b_i2^i = (2^N-1)C_uV_x + V_x C_p + (V_x -
V_o)2^NC_u
\] therefore \[
-V_o = \frac{2^N
C_u}{\frac{(2^{N+1}-1)C_u+C_p}{A}+2^NC_u}\sum_{i=0}^{N-1}b_i\left(2^i\frac{V_R}{2^N}\right)\approx
\sum_{i=0}^{N-1}b_i\left(2^i\frac{V_R}{2^N}\right)
\]
Midscale (MSB Transition) often is the largest DNL
error
\(C_4\) and \(C_1+C_2+C_3\) are independent (can't cancel
out) and their variance is two largest (\(16\sigma_u^2\), \(15\sigma_u^2\), ), the total standard
deviation is \(\sqrt{16\sigma_u^2+15\sigma_u^2}=\sqrt{31}\sigma_u\)
CDAC with Custom MoM
P. J. A. Harpe et al., "A 26 μ W 8 bit 10 MS/s Asynchronous
SAR ADC for Low Energy Radios," in IEEE Journal of Solid-State
Circuits, vol. 46, no. 7, pp. 1585-1595, July 2011 [https://sci-hub.ru/10.1109/JSSC.2011.2143870]
P. Harpe, "A Compact 10-b SAR ADC With Unit-Length Capacitors and a
Passive FIR Filter," in IEEE Journal of Solid-State Circuits,
vol. 54, no. 3, pp. 636-645, March 2019 [https://sci-hub.ru/10.1109/JSSC.2018.2878830]
CR Switching Schemes
Hariprasath, V., Jon Guerber, Seunghoon Lee and Un-Ku Moon. “Merged
capacitor switching based SAR ADC with highest switching
energy-efficiency.” Electronics Letters 46 (2010): 620-621. [https://sci-hub.ru/10.1049/EL.2010.0706]
Y. Zhu et al., "A 10-bit 100-MS/s Reference-Free SAR ADC in
90 nm CMOS," in IEEE Journal of Solid-State Circuits, vol. 45,
no. 6, pp. 1111-1121, June 2010 [https://sci-hub.ru/10.1109/JSSC.2010.2048498]
C. -C. Liu, S. -J. Chang, G. -Y. Huang and Y. -Z. Lin, "A 10-bit
50-MS/s SAR ADC With a Monotonic Capacitor Switching Procedure," in
IEEE Journal of Solid-State Circuits, vol. 45, no. 4, pp.
731-740, April 2010 [https://sci-hub.ru/10.1109/JSSC.2010.2042254]
Rabuske, Taimur & Fernandes, Jorge. (2017). "Review of SAR ADC
Switching Schemes" Charge-Sharing SAR ADCs for Low-Voltage Low-Power
Applications. 10.1007/978-3-319-39624-8. [https://sci-hub.ru/10.1007/978-3-319-39624-8_3]
Ramkaj, A.T.; Pelgrom, M.J.M.; Steyaert, M.S.J.; Tavernier, F.
Multi-Gigahertz Nyquist Analog-to-Digital Converters: Architecture
and Circuit Innovations in Deep-Scaled CMOS and FinFET
Technologies; Springer International Publishing: Berlin/Heidelberg,
Germany, 2023.
Charge Redistribution (CR) Switching Schemes:
Conventional CR
switching scheme: No CM variation
Monotonic switching
scheme: Single-ended operation with CM varies
50%VFS
VCM-based switching
scheme, a.k.a merged capacitor switching
(MCS): No CM variation
Conventional (trial-and-keep) switches the trial
capacitor in first, then compares. If the result says the trial
overshot, it switches that capacitor back out and inserts the next one.
Bits are stored as which capacitors stay connected to \(V_{REF}\)
Monotonic (compare-then-move) compares first, with
no trial applied. Then it discharges the next capacitor (from \(V_{REF}\) to ground) on whichever side is
higher. That move is never undone; it simply sets up the next
comparison
Conventional Switching
bottom-plate
sampling
The comparator common mode is \(V_{CM} +
V_{REF}/2 − V_{in,cm}\)
Node voltages after the MSB switching
Each array has \(8C\) in total
(\(4C + 2C + C + C\), counting the
dummy unit cap). During sampling, the top plates sit at \(V_{CM}\) and the bottom plates sit at the
input.
For the top (P) array, the charge is \(8C(V_{CM} − V_{INP})\). Then \(4C\) is switched to \(V_{REF}\) and \(4C\) to ground. Charge conservation
gives:
You can't drop the input common-mode term. The figure assumes the
top-plate sampling voltage equals the input common mode (\(V_{CM} = V_{in,cm} = 0.5\) V). Those two
cancel, and the comparator common mode is simply \(V_{REF}/2 = 0.5\) V. That is why both
waveforms start at 0.5 V during sampling and converge back to 0.5 V.
Check against the plot
With \(V_{INP} = 0.9\) V, \(V_{INN} = 0.1\) V, \(V_{CM} = 0.5\) V, and \(V_{REF} = 1\) V:
For simplicity, the final term \(\color{blue}(b[0]-1)\cdot 1\text{LSB}\) of
the general equation \(D_{out} = s(M) +
\sum_{i=1}^{M-1}(2\cdot b[i] - 1)\times s(i) + (b[0] -1)\cdot
\text{1LSB}\) is approximated as \(\color{red}(2b[0]
-1)\cdot\frac{\text{1LSB}}{2}\)
During bit cycling, \(V_y - V_x \to
0\), i.e. \[
2 \textcolor{red}{\sum b_i \Delta V_i } -
\textcolor{blue}{V_{REF}}\qquad \Longrightarrow \qquad V_{INP} - V_{INN}
\] After N = 8 bits: \[
V_{INP} - V_{INN} = V_{REF}\left(2\sum_{i=1}^{8} b_i\,2^{-i} - 1\right)
+ \varepsilon,
\qquad 0 \le \varepsilon < \frac{2V_{REF}}{2^8} = \mathrm{1 LSB}
\] For the complete 8-bit conversion shown in Fig. 3.3 \[
2\times
\left(\frac{1}{2}\times1+\frac{1}{4}\times1+\frac{1}{8}\times1+\frac{1}{16}\times0+\frac{1}{32}\times0+\frac{1}{64}\times1+\frac{1}{128}\times1+\frac{1}{256}\times0\right)-1
= \frac{204}{256}
\] i.e. \[
\left|\frac{204}{256} - 0.8\right| \qquad \lt \qquad \mathrm{1LSB}
\]
so the relationship between Conventional and monotonic result \[
V_\text{monotonic} = V_\text{Conventional} + \color{red}\frac{\mathrm{1
LSB}}{2}
\] For the complete 8-bit conversion of monotonic scheme shown in
Fig. 3.4 \[
\frac{1}{2}+\frac{1}{4}+\frac{1}{8}-\frac{1}{16}-\frac{1}{32}+\frac{1}{64}+\frac{1}{128}-\frac{1}{256}
= \frac{204}{256} + \color{red}\frac{1}{256}
\]
3-bit conversion of monotonic switching scheme as
Fig.3.4 drawn
where \[
b_k = \begin{cases} 1 & \text{if } V_x^{(k-1)} \ge V_y^{(k-1)} \\ 0
& \text{otherwise } \end{cases}
\] During bit cycling, \(V_y - V_x \to
0\), i.e. \[
\textcolor{red}{\sum (2b_i-1) \Delta V_{\downarrow,i} + \sum
(2b_i-1) \Delta V_{\uparrow,i}} \qquad \Longrightarrow \qquad V_{INP}
- V_{INN}
\] When \(V_{CM}\) is precisely
half of \(V_{REF}\)\[
\Delta V_{\downarrow,i} = \Delta V_{\uparrow,i} =
\frac{V_{REF}}{2^{i+1}}
\] then \[
V_{INP} - V_{INN} \approx \sum (2b_i-1) \frac{V_{REF}}{2^{i}}
\]
CDAC with constant
common-mode voltage
Synchronous SAR ADC
It also divides a full conversion into several comparison stages in a
way similar to the pipeline ADC, except the algorithm is
executed sequentially rather than in parallel
as in the pipeline case.
However, the sequential operation of the SA algorithm has
traditionally been a limitation in achieving high-speed
operation
a clock running at least \((N + 1) \cdot
F_s\) is required for an \(N\)-bit converter with conversion rate of
\(F_s\)
every clock cycle has to tolerate the worst case comparison
time
every clock cycle requires margin for the clock jitter
The power and speed limitations of a synchronous SA design comes
largely from the high-speed internal clock
Mike Shuo-Wei Chen and R. W. Brodersen, "A 6-bit 600-MS/s 5.3-mW
Asynchronous ADC in 0.13-μm CMOS," in IEEE Journal of Solid-State
Circuits, vol. 41, no. 12, pp. 2669-2680, Dec. 2006 [pdf,
slides]
—. "Asynchronous SAR ADC: Past, Present and Beyond"
H. Karrari, P. Andreani and S. Tan, "Asynchronous vs Synchronous SAR
ADCs – Performance Beyond Nominal Speed," 2024 19th Conference on
Ph.D Research in Microelectronics and Electronics (PRIME), Larnaca,
Cyprus, 2024, pp. 1-4, doi: 10.1109/PRIME61930.2024.10559690.
The comparator itself trigger the next bit-conversion cycle as soon
as the present bit decision has been taken
The maximum resolving time reduction between synchronous and
asynchronous case is two fold
comparator metastable state
when the input is sufficiently small. The time needed for
the comparator outputs to fully resolve may take arbitrarily
long
In this case, the ready signal generator should still set the
flag and the decision result is simply taken from the previous
value stored in the SR latch
both outputs (\(Q_p\) and \(Q_n\)) will drop together, NAND is
inverter actually
The transition point of this NAND gate is skewed to
eliminate metastability issues arising when the input differential
voltage level is small (comparator)
Redundancy with Radix
Kuttner, Franz. "A 1.2V 10b 20MSample/s non-binary successive
approximation ADC in 0.13/spl mu/m CMOS." 2002 IEEE International
Solid-State Circuits Conference. Digest of Technical Papers (Cat.
No.02CH37315) 1 (2002): 176-177 vol.1. [https://sci-hub.jp/10.1109/ISSCC.2002.992993]
M. Hesener, T. Eicher, A. Hanneberg, D. Herbison, F. Kuttner and H.
Wenske, "A 14b 40MS/s Redundant SAR ADC with 480MHz Clock in 0.13pm
CMOS," 2007 IEEE International Solid-State Circuits Conference.
Digest of Technical Papers, San Francisco, CA, USA, 2007, pp.
248-600 [https://sci-hub.ru/10.1109/ISSCC.2007.373387]
Tomohiko OGAWA et al., SAR ADC Algorithm with Redundancy and
Digital Error Correction, IEICE Transactions on Fundamentals of
Electronics, Communications and Computer Sciences, 2010, Volume E93.A,
Issue 2, Pages 415-423 [https://sci-hub.ru/10.1587/TRANSFUN.E93.A.415]
C. -C. Liu et al., "A 10b 100MS/s 1.13mW SAR ADC with
binary-scaled error compensation," 2010 IEEE International
Solid-State Circuits Conference - (ISSCC), San Francisco, CA, USA,
2010, pp. 386-387 [https://sci-hub.ru/10.1109/ISSCC.2010.5433970]
—, “Design of High-Speed Energy-Efficient Successive-Approximation
Analog-to-Digital Converters,” Ph.D. dissertation, Dept. Elect. Eng.,
National Cheng Kung University, Tainan, Taiwan, R.O.C., 2010.
—, "A 10 bit 320 MS/s Low-Cost SAR ADC for IEEE 802.11ac Applications
in 20 nm CMOS," in IEEE Journal of Solid-State Circuits, vol.
50, no. 11, pp. 2645-2654, Nov. 2015 [https://sci-hub.ru/10.1109/JSSC.2015.2466475]
W. Liu, P. Huang and Y. Chiu, "A 12-bit, 45-MS/s, 3-mW Redundant
Successive-Approximation-Register Analog-to-Digital Converter With
Digital Calibration," in IEEE Journal of Solid-State Circuits, vol. 46,
no. 11, pp. 2661-2672, Nov. 2011 [https://sci-hub.ru/10.1109/JSSC.2011.2163556]
Albert H. Chang, Hae-Seung Lee, and Duane S. Boning. 2011. Redundancy
in SAR ADCs. In Proceedings of the 21st edition of the great lakes
symposium on Great lakes symposium on VLSI (GLSVLSI '11). Association
for Computing Machinery, New York, NY, USA, 283–288. [https://dl.acm.org/doi/10.1145/1973009.1973066]
—, “Low-power high-performance SAR ADC with redundancy and digital
background calibration,” Ph.D. dissertation, Dept. Elect. Eng. Comput.
Sci., Massachusetts Institute of Technology, Cambridge, MA, USA, 2013.
[Online]. [https://dspace.mit.edu/bitstream/handle/1721.1/82177/861702792-MIT.pdf]
Krämer, M. et al. (2015) High-resolution SAR A/D converters with
loop-embedded input buffer. dissertation. Available at: [http://purl.stanford.edu/fc450zc8031].
After each DAC switching, the effective input range is
reduced by a factor of 2
Kuttner ISSCC'02: non-binary SAR
ADC
the input range is reduced by a factor smaller than 2 per
each bit cycle
Liu ISSCC'10: binary-scaled
CDAC
Max tolerance of comparator offset is \(\pm
V_{FS}/4\)
\(b_j\) error is \(\pm 1\)
\(b_{j+1}\) error is \(\pm 2\) , wherein \(b_{j+1}\): \(0\to
2\) or \(1\to -1\)
i.e. complementary analog and digital errors cancel each other, \(V_o +\Delta V_{o}\) should be in
over-/under-range comparators (\(-V_{FS}/2 \sim 3V_{FS}/2\))
Note: when \(\sum_{i=0}^{M-1} W_i = (2^N - 1)\) LSB, the
constant term simplifies to \(\frac{1}{2}\left[\sum_{i=0}^{M-1} W_i + W_0\right]
= 2^{N-1}\) LSB, so
Binary with
Error Compensation a.k.a Binary-Scaled Error
Compensation
this is the monotonic
scheme
Each comparison is followed by switching one capacitor (\(V_{REF} \to \text{ground}\)) on whichever
side is higher. That moves the differential DAC voltage by \(\pm w\,u\), where \(w\) is that capacitor's size in unit caps
(\(u\) below). The order is:
B1 → C1 (256), B2 → C2 (128), B3 → C3 (64)
B3C → C3C (64)
B4 → C4 (32), B5 → C5 (16), B6 → C6 (8)
B6C → C6C (8)
B7 → C7 (4), B8 → C8 (2), B9 → C9 (1)
B9C → C9C (1)
B10 → nothing (last
decision)
That is 13 decisions, giving the "13b redundant code."
Let one unit cap move the differential by \(u = V_{REF}\cdot C_u/C_{total}\), and let
\(w_k = 2^{9-k}\) be the size of \(C_k\). After the last move (C9C) the
residue is within \(\pm u\); B10
resolves its sign, so the reconstruction error is within \(\pm u/2\): \[
\frac{V_{in}}{u} \approx \underbrace{\sum_{k=1}^{9}
(2B_k-1)\,w_k}_{\text{main caps}} \;+\; \underbrace{(2B_{3C}-1)\,64 +
(2B_{6C}-1)\,8 + (2B_{9C}-1)\,1}_{\text{compensation caps}} \;+\;
(2B_{10}-1)\tfrac12
\]
The 10-bit input range is \(\color{blue}\pm512u\): \(C_1\sim C_9\) span \(\pm511u\) and B10 resolves the last \(\pm u\). The compensation caps add \(\pm73u\) of over-range (raw code \(-73\dots1096\)), which the DEC clips. Since
\(C_{total}\approx584\,C_u\), the full
scale is \(\pm\tfrac{512}{584}V_{REF}\approx\pm0.88\,V_{REF}\)
(ignoring parasitics)
So the output code, with intentional offset \(511.5\)\[
D_{out} \approx \frac{V_{in}}{u} + \color{red}\left(511+\tfrac12\right)
\] which maps the input onto \(0 \dots
1023\)
For the \(n\)th output bit, once a
decision is made, the next decision level will either move up or down by
the step size of \(s(n − 1)\)
If this decision is erroneous, then the sum of the follow-on step
sizes, \(s(n − 2)\), \(s(n − 3)\), ..., \(s(1)\), must be large enough and exceed the
value of the current step size to counteract this mistake
The exceeded amount is the tolerance window for that decision
level
J. Craninckx and G. van der Plas, "A 65fJ/Conversion-Step 0-to-50MS/s
0-to-0.7mW 9b Charge-Sharing SAR ADC in 90nm Digital CMOS," 2007
IEEE International Solid-State Circuits Conference. Digest of Technical
Papers, San Francisco, CA, USA, 2007, pp. 246-600 [https://sci-hub.jp/10.1109/ISSCC.2007.373386]
V. Giannini, P. Nuzzo, V. Chironi, A. Baschirotto, G. Van der Plas
and J. Craninckx, "An 820μW 9b 40MS/s Noise-Tolerant Dynamic-SAR ADC in
90nm Digital CMOS," 2008 IEEE International Solid-State Circuits
Conference - Digest of Technical Papers, San Francisco, CA, USA,
2008, pp. 238-610 [https://sci-hub.jp/10.1109/ISSCC.2008.4523145]
Rabuske, Taimur & Fernandes, Jorge. (2016). Charge-Sharing SAR
ADCs for Low-Voltage Low-Power Applications.
10.1007/978-3-319-39624-8.
Charge-based reference
signal
The trick is to analyze it in the charge domain
rather than the voltage domain
polarity
switching in charge-sharing SAR capacitor arrays
Each DAC capacitor C_i has two plates, A and B, and typically six
switches:
Two precharge switches: A to Vref and B to GND,
closed during the sampling phase.
Two "straight" switches: A to P and B to N.
Two "crossed" switches: A to N and B to P
With \(C_A = C_B = C_s\) and \(\Delta V_C = V_{CP} - V_{CN}\): \[\begin{align}
Q_A' &= V_A C_s + \Delta V_C \cdot C_C = V_A' C_s +
(V_A' - V_B')C_C \\
Q_B' &= V_B C_s - \Delta V_C \cdot C_C = V_B' C_s -
(V_A' - V_B')C_C
\end{align}\] then \[
V_A' - V_B' = \frac{\textcolor{red}{C_s}\,(V_A - V_B) +
\textcolor{red}{2C_C}\,\Delta V_C}{C_s + 2C_C}
\qquad
\frac{V_A' + V_B'}{2} = \frac{V_A + V_B}{2}
\] This confirms the earlier claim: a floating
capacitor\(C_C\) across \(P\) and \(N\) acts like \(2C_C\) in the single-ended picture. A
straight connection has \(\Delta V_C =
+V_\text{ref}\) (add), and a flipped connection has \(\Delta V_C = -V_\text{ref}\)
An ideal floating capacitor injects zero common-mode
charge
Aspect
Charge redistribution
Charge sharing
Reference during bit cycles
Connected, must settle each cycle
Disconnected, only precharges during sampling
Reference current
Signal-dependent
Constant (≈ C_DAC·Vref per conversion)
Reference buffer demand
High, especially at high fs
Relaxed; a large decoupling cap often suffices
Signal at comparator
Full swing
Attenuated by C_s / (C_s + C_DAC + C_p)
Full-scale / gain
Set by Vref
Set by Vref × capacitor ratio, sensitive to parasitics
Leakage sensitivity
Low
Higher, since precharged caps float during conversion
Typical resolution
Up to 16–18 bit
Usually ≤ 10–11 bit
reference
Andrea Baschirotto, ISSCC2009 T6: SAR ADCs
Pieter Harpe, ISSCC 2016 Tutorial: "Basics of SAR ADCs Circuits &
Architectures"
L. Jie et al., "An Overview of Noise-Shaping SAR ADC: From
Fundamentals to the Frontier," in IEEE Open Journal of the Solid-State
Circuits Society, vol. 1, pp. 149-161, 2021 [pdf]
Walden FoM unit: J/conv-step [joules per
conversion]
"Conversion-step" in the Walden FoM doesn't mean a physical operation
like a comparator decision or a clock cycle. It means one
quantization level, i.e., one effective LSB step out of
the 2ENOB levels the converter can distinguish
measured in digital domain:
long-term averages of ADC out
If the input has a nonzero mean, the output average also contains the
signal’s DC component, so it does not identify offset alone
corrected in analog domain:
ADC dynamic range reduction due to the offset, which
holds if we force the offset of each channel to zero rather than make
the offsets of different channels equal
The offset-correction DAC here is a switched-capacitor DAC. Its
digital code selects which capacitor bottom plates switch between ground
and \(V_{\mathrm{REF}}\)
The op-amp holds the summing node approximately at virtual
ground. The injected charge is balanced through feedback
capacitor \(C_2\), causing
\(V_{\mathrm{res}}\) to
change.
For an ideal op-amp, the correction-induced output step is
the SAR DAC gain decreases, whereas the ADC’s
output-code-per-volt gain increases. In the shown circuit, the
sampling switch directly sets \(V_X=V_{\mathrm{in}}\), so the sampled input
is not attenuated. Smaller DAC steps mean more code is needed to balance
the same input: \[
\boxed{\frac{G_{\mathrm{ADC}}}{G_{\mathrm{ADC,ideal}}} =\frac{1}{\alpha}
=1+\frac{C_{\mathrm{CALIB}}}{C_{\mathrm{SAR}}}}
\]
The LSB determines the ADC code resolution, but the minimum reliably
detectable input voltage is determined by the total noise floor,
including quantization noise
ADC quantization noise is considered a "deterministic error" rather
than random thermal noise because it correlates with the input signal,
creating a saw-tooth error waveform rather than random Gaussian noise.
While it cannot be reduced by averaging a static signal, it
can be reduced through averaging if the signal is
oversampled, or if dither (random noise) is added to decorrelate it.
Quantization is NOT Noise
N. Blachman, "The intermodulation and distortion due to quantization
of sinusoids," in IEEE Transactions on Acoustics, Speech, and Signal
Processing, vol. 33, no. 6, pp. 1417-1426, December 1985 [https://sci-hub.st/10.1109/TASSP.1985.1164729]
N. Blachman, "The intermodulation and distortion due to quantization
of sinusoids," in IEEE Transactions on Acoustics, Speech, and Signal
Processing, vol. 33, no. 6, pp. 1417-1426, December 1985
The quantization noise is an infinite sum of input signal
odd harmonics, where the amplitude of the harmonics is
determined by a sum of a Bessel function
"Quantization noise is white", because for a high
number of bits, it looks white in the FFT
The quantization noise is odd harmonics of the input
signal [Gist
link]
In the power domain, \(\color{red}v_{nS,RMS}/3 \lt
\sigma_{q,RMS}\) ensures that sampling noise power is nearly
an order of magnitude smaller than the quantization noise
ADC SNR & clock jitter
Akkaya, A. (2021). High-Speed ADC Design and Optimization for
Wireline Links (Publication No. 8453) [PhD thesis, EPFL; Supervised by
Y. Leblebici]. [https://doi.org/10.5075/epfl-thesis-8453]
Unlike the quantization noise and the thermal noise, the impact of
the clock jitter on the ADC performance depends on the input signal
properties like its PSD
The error between the ideal sampled signal and the
sampling with clock jitter can be treated as noise and it results
in the degradation of the SNR of the ADC
K. Tyagi and B. Razavi, "Performance Bounds of ADC-Based Receivers
Due to Clock Jitter," in IEEE Transactions on Circuits and Systems
II: Express Briefs, vol. 70, no. 5, pp. 1749-1753, May 2023 [https://www.seas.ucla.edu/brweb/papers/Journals/KT_TCAS_2023.pdf]
N. Da Dalt, M. Harteneck, C. Sandner and A. Wiesbauer, "On the jitter
requirements of the sampling clock for analog-to-digital converters," in
IEEE Transactions on Circuits and Systems I: Fundamental Theory and
Applications, vol. 49, no. 9, pp. 1354-1360, Sept. 2002 [https://sci-hub.se/10.1109/TCSI.2002.802353]
M. Shinagawa, Y. Akazawa and T. Wakimoto, "Jitter analysis of
high-speed sampling systems," in IEEE Journal of Solid-State Circuits,
vol. 25, no. 1, pp. 220-224, Feb. 1990 [https://sci-hub.se/10.1109/4.50307]
Akkaya, A. (2021). High-Speed ADC Design and Optimization for
Wireline Links (Publication No. 8453) [PhD thesis, EPFL; Supervised by
Y. Leblebici]. [https://doi.org/10.5075/epfl-thesis-8453]
Boris Murmann ISSCC 2022 SC1: Introduction to ADCs/DACs: Metrics,
Topologies, Trade Space, and Applications [pdf]
S. Kim, K. -Y. Lee and M. Lee, "Modeling Random Clock Jitter Effect
of High-Speed Current-Steering NRZ and RZ DAC," in IEEE Transactions
on Circuits and Systems I: Regular Papers, vol. 65, no. 9, pp.
2832-2841, Sept. 2018 [https://sci-hub.se/10.1109/TCSI.2018.2821198]
Martin Clara. High-Performance D/A-Converters - Application to
Digital Transceivers, 2013 [pdf]
Chun-Hsien Su (蘇純賢). Design of Oversampled Sigma-Delta Data
Converters. July, 2006 [pdf]
ampling Jitter Effects for ADC/DAC
In both DAC or ADC cases, doubling the timing jitter doubles the
noise level
Also, doubling the frequency or amplitude doubles the jitter induced
noise - SNR is not improved
The quantization noise power \(P_Q\)
for a uniform quantizer with step size \(\Delta\) is given by \[
P_Q = \frac{\Delta ^2}{12}
\] For a full-scale sinusoidal input signal with an amplitude
equal to \(V_{FS}/2\), the input signal
is given by \(x(t) =
\frac{V_{FS}}{2}\sin(\omega t)\)
Then input signal power \(P_s\) is
\[
P_s = \frac{V_{FS}^2}{8}
\] Therefore, the signal-to-quantization noise ratio
(SQNR) is given by \[
\text{SQNR} = \frac{P_s}{P_Q} =
\frac{V_{FS}^2/8}{\Delta^2/12}=\frac{V_{FS}^2/8}{V_{FS}^2/(12\times
2^{2N})} = \frac{3\times 2^{2N}}{2}
\] where \(N\) is the number of
quantization bits
When represented in dBs \[
\text{SQNR(dB)} = 10\log(\frac{P_s}{P_Q}) = 10\log(\frac{3\times
2^{2N}}{2})= 20N\log(2) + 10\log(\frac{3}{2})= 6.02N + 1.76
\]
the maximum achievable SNR of N-bit ADC — theoretical SNR limit
Dan Boschen, GRCon25: Quantifying Signal Quality: Practical Tools for
High-Fidelity Waveform Analysis
Two ways to deal with spectral leakage: Ensure integer
number of periods or
Windowing
Coherent Sampling
Choosing M/N non-prime repeats the signal quantization periodically
and fewer quantization steps are measured. The quantization repeats
periodically and creates a line spectrum that can obscure real frequency
lines (e.g. the red lines in the images below, created by
non-linearities of the ADC).[https://www.dsprelated.com/thread/469/coherent-sampling-very-brief-and-simple]
N = 2048; cycles = 67; fs = 1000; fx = fs*cycles/N; LSB = 2/2^10; %generate signal, quantize (mid-tread) and take FFT xc = cos(2*pi*fx/fs*[0:N-1]); x = round(xc/LSB)*LSB; s = abs(fft(x)); s = s(1:end/2)/N*2; % calculate SNR sigbin = 1 + cycles; noise = [s(1:sigbin-1), s(sigbin+1:end)]; snr = 10*log10( s(sigbin)^2/sum(noise.^2) );
% frequency vector f = [0:N/2-1]/N; subplot(1,3,1) stem(f, s); % some FFT bins for "noise" to be exactly zero, empty plot subplot(1,3,2) plot(f, 20*log10(s))
% Use a small offset to avoid -Inf subplot(1,3,3) plot(f, 20*log10(s + 1e-6)) ylim([-120, 0])
Periodic Quantization Noise if N and cycles
are not mutually prime, i.e.
cycles=64 then GCD(2048, 64)=64, then \(N_R' = 2048/64=32\), so quantization
noise manifests as odd harmonics\(n/32\mid n = 2k + 1, k \in \mathbb{Z}\)
because of Half Wave Symmetry
import numpy as np from matplotlib.ticker import EngFormatter
defcoherent_sampling(fin, Fs, N): """ Returns the adjusted input frequency for coherent sampling. Parameters: fin : input frequency (Hz) Fs : sample rate (Hz) N : number of samples Returns: Adjusted fin (Hz) such that M/N = fin/Fs, where M is the nearest prime to N*fin/Fs. """ x = int(np.ceil(N * fin / Fs)) primes = [n for n inrange(2, x*2) ifall(n % i for i inrange(2, n))] M = primes[np.argmin(np.abs(np.array(primes) - x))] return Fs * M / N
D. Pfaff et al., "7.3 A 224Gb/s 3pJ/b 40dB Insertion Loss
Transceiver in 3nm FinFET CMOS," 2024 IEEE International Solid-State
Circuits Conference (ISSCC), San Francisco, CA, USA, 2024 [https://iccircle.com/static/upload/img20240529101747.pdf]
—, "A 224Gb/s 3pJ/bit 42dB Insertion Loss Post-FEC Error Free
Transceiver in 3-nm FinFET CMOS (Invited)," 2025 IEEE Custom
Integrated Circuits Conference (CICC), Boston, MA, USA, 2025, pp.
1-8, doi: 10.1109/CICC63670.2025.10983461.
E. Swindlehurst et al., "An 8-bit 10-GHz 21-mW Time-Interleaved SAR
ADC With Grouped DAC Capacitors and Dual-Path Bootstrapped Switch," IEEE
Journal of Solid-State Circuits, vol. 56, no. 8, pp. 2347-2359, 2021,
[https://sci-hub.se/10.1109/JSSC.2021.3057372]
Tracking nonidealities
Tracking Settling Accuracy
Finite Acquisition Time - Consider a sinusoidal
input
The relation \(V_\text{out}(s) =
H(s)\,V_\text{in}(s)\) is the zero-state
response only. The transfer function is defined under the
assumption \(V_\text{out}(0^-) = 0\),
so you can't recover the natural response from it. You
have to go back one step, to the differential
equation.
Transform with the full derivative rule \(\mathcal{L}\{\dot{V}_\text{out}\} = s
V_\text{out}(s) - V_0\), where \(V_0
\equiv V_\text{out}(0^-)\) (the capacitor voltage is continuous,
so \(0^-\) and \(0^+\) agree): \[
\tau\big(s V_\text{out}(s) - V_0\big) + V_\text{out}(s) = V_\text{in}(s)
\]
\[
\boxed{\;V_\text{out}(s)
= \underbrace{\frac{V_\text{in}(s)}{\tau s + 1}}_{\text{zero-state}}
+ \underbrace{\frac{\tau V_0}{\tau s + 1}}_{\text{zero-input}}
= H(s)\,V_\text{in}(s) + \frac{V_0}{s + \frac{1}{\tau}}\;}
\]
Then the Complete Laplace Expression is \[
\boxed{V_\text{out}(s) = -\frac{\cos(\theta - \phi)}{\sqrt{\omega^2\tau
^2 +1}} \frac{1}{s+\frac{1}{\tau}} + \frac{1}{\sqrt{\omega^2\tau ^2
+1}}\frac{s\cos(\theta - \phi) - \omega \sin(\theta -
\phi)}{s^2+\omega^2} + \frac{V_\text{out}(0)}{s + \frac{1}{\tau}} }
\] The Total Time-Domain Solution\[
\boxed{V_{\text{out}}(t) =
\underbrace{V_{\text{out}}(0)e^{-\frac{t}{\tau}}}_{\text{Pure Natural
Response}} + \underbrace{\frac{1}{\sqrt{\omega^2\tau^2 + 1}} \left[
\cos(\omega t + \theta - \phi) - \cos(\theta - \phi) e^{-\frac{t}{\tau}}
\right]}_{\text{Forced Response (Transient + Steady-State)}}
}
\] And grouped by transient vs. steady-state components: \[
\boxed{V_{\text{out}}(t) = \underbrace{\left[ V_{\text{out}}(0) -
\frac{\cos(\theta - \phi)}{\sqrt{\omega^2\tau^2 + 1}} \right]
e^{-\frac{t}{\tau}}}_{\text{Total Transient Response}} +
\underbrace{\frac{\cos(\omega t + \theta - \phi)}{\sqrt{\omega^2\tau^2 +
1}}}_{\text{Steady-State Response}}}
\]
Tracking Nonlinearity
Wei Yu, Subhajit Sen and B. H. Leung, "Distortion analysis of MOS
track-and-hold sampling mixers using time-varying Volterra series," in
IEEE Transactions on Circuits and Systems II: Analog and Digital Signal
Processing, vol. 46, no. 2, pp. 101-113, Feb. 1999 [https://sci-hub.ru/10.1109/82.752910]
It divides the process into several comparison stages, the number of
which is proportional to the number of bits
Due to the pipeline structure of both analog and digital signal path,
inter-stage residue amplification is needed which
consumes considerable power and limits high speed operation
Non-Flip-Around &
Flip-Around Amplifier
Non-Flip-Around
Amplifier
The amplifier's differential input stays approximately zero after
settling; its individual input voltages need not be constant
With matched capacitors, their common-mode contributions cancel when
we subtract the two charge-conservation equations
Flip-Around
Amplifier
Multiplying DACs (MDAC)
Connection
Sampling phase
Amplification phase
(V_{}) to left plate of (C_S)
Closed
Open
Left plate of (C_S) to ground
Open
Closed
Node (x) to ground
Closed
Open
Output to ground
Closed
Open
Phase 1: sample the input
\(C_S\) has \(V_{\text{in}}\) on its left plate and 0 V
on its right plate, so it stores the input as
charge.
Both ends of \(C_F\) are grounded,
resetting its voltage to zero.
The small ADC samples the same input and determines a digital
code.
At the end of sampling:
\[
Q_{x,\text{sample}}=-C_SV_{\text{in}}
\]
Phase 2: subtract and amplify
Switching (C_S)’s left plate from (V_{}) to zero disturbs node
(x).
Switching the DAC bottom plate from zero to (V_D) produces an
opposing disturbance.
The amplifier changes (V_{}), through (C_F), to balance the
remaining charge.
After amplification settles, both ends of \(C_S\) are approximately
zero:
Samuel Palermo, ISSCC 2018 T10: ADC-Based Serial Links: Design and
Analysis
Jan Mulder Broadcom. ISSCC2015 T5: High-Speed Current-Steering
DACs
Zhang, Milin, Zhihua Wang, Jan van der Spiegel and Franco Maloberti.
"Advanced Tutorial on Analog Circuit Design." (2023)
V. Chen, "Tutorial: High-Speed Analog-to-Digital Converters,"
2025 IEEE International Solid-State Circuits Conference
(ISSCC), San Francisco, CA, USA, 2025, pp. 1-1, doi:
10.1109/ISSCC49661.2025.11076112.
S. Su, "Principles and Practices of High-Speed DAC Design: From
Conversion Fundamentals to Layout-Aware Implementation," in IEEE
Solid-State Circuits Magazine, vol. 18, no. 3, pp. 26-43, Summer
2026, doi: 10.1109/MSSC.2026.3704105
Noise of ref MOS is amplified by the current mirror's gain
Due to M0 is biased by current source, we probe gate voltage instead
of current
result browser of instance show noise
contribution, which is same with noise summary
current mirror with source
follower
source follower alleviate gate leakage impact on reference
current
Constant-gm Bias Circuit
S. Pavan, "Systematic Development of CMOS Fixed-Transconductance Bias
Circuits," in IEEE Transactions on Circuits and Systems II: Express
Briefs, vol. 69, no. 5, pp. 2394-2397, May 2022 [https://sci-hub.ru/10.1109/TCSII.2022.3158358]
[slides]
—, ISSCC2026: Circuit Insights: Voltage and Current Reference
Generation
aka. Beta-multiplier
reference, Fixed-transconductance bias
current
\(I_\text{out}\) is
PTAT in case temperature coefficient of \(R_s\) is less than that of \(\mu_n\)
Both the circuits incorporate positive feedback
left circuit has a loop gain less than one,
indicating a stable operating point
right circuit has a loop gain greater than unity
and is unsuitable
Body effect of M2
Boris Murmann, Systematic Design of Analog Circuits Using
Pre-Computed Lookup Tables
S. Pavan, "Systematic Development of CMOS Fixed-Transconductance Bias
Circuits," in IEEE Transactions on Circuits and Systems II: Express
Briefs, vol. 69, no. 5, pp. 2394-2397, May 2022
S. Pavan, "A Fixed Transconductance Bias Circuit for CMOS Analog
Integrated Circuits", IEEE International Symposium on Circuits and
Systems, ISCAS 2004, Vancouver , May 2004
Why MOS in saturation ?
\(g_m\), \(g_\text{ds}\) at fixed \(V_\text{GS}\)
\(g_{ds}\) is constant in saturation
region
in triode region \[
g_{ds} = \mu_nC_{ox}\frac{W}{L}(V_{GS}-V_{TH}-V_{DS})
\]
Interestingly, \(g_m\) in the
saturation region is equal to the inverse of \(R_\text{on}\) in the deep triode
region.
\(g_m\), \(g_\text{ds}\) at fixed \(I_d\), \(V_G\)
In triode region\[
I_D =
\frac{1}{2}\mu_nC_{ox}\frac{W}{L}[2(V_{GS}-V_{TH})V_{DS}-V_{DS}^2]
\] where \(I_D\) and \(V_G\) is fixed
Then \(V_S\) can be expressed with
\(V_D\), that is \[
V_S = V_{GT} - \sqrt{(V_{GT}-V_D)^2+V_{dsat}^2}
\] where \(V_{GT}=V_G-V_{TH}\),
\(V_{dsat}\) is \(V_{DS}\) saturation voltage \[
g_m =
\mu_nC_{ox}\frac{W}{L}\left(V_D-V_{GT}+\sqrt{(V_{GT}-V_D)^2+V_{dsat}^2}\right)
\] Then \[
\frac{\partial g_m}{\partial V_D} \propto 1 -
\frac{V_{GT}-V_D}{\sqrt{(V_{GT}-V_D)^2+V_{dsat}^2}} \gt 0
\]
Another solution is to view the mirror branch as a
whole
In source branch self mismatch \(\sigma_{vth,src} =
\frac{A_{vt}}{\sqrt{2WL}}\) and mirror branch self-mismatch \(\sigma_{vth,mir} =
\frac{A_{vt}}{\sqrt{2kWL}}\), then mutual mismatch is \[
\sigma_{vth} = \sqrt{\sigma_{vth,src}^2 + \sigma_{vth,mir}^2} =
\sigma_{vth,src}\sqrt{1+\frac{1}{k}}
\] mirror current variation \(\sigma_{I_k} = k g_m \sigma_{vth}\),
relative variation of mirror current is \[
\frac{\sigma_{I_k}}{I_k} = \frac{k g_m \sigma_{vth}}{kI} = \frac{g_m
\sigma_{vth,src}}{I}\sqrt{1+\frac{1}{k}}=\color{red}\frac{\sigma_{I}}{I}\sqrt{1+\frac{1}{k}}
\]
Biasing
current source and global variation Monte Carlo
iwl: biased by mirror
iwl_ideal: biased by vdc source, whose
value is typical corner
For local variation, constant voltage bias
(vb_const in schematic) help reduce variation from \(\sqrt{2}\Delta V_{th}\) to \(\Delta V_{th}\)
For global variation, all device have same
variation, mirror help reduce variation by sharing same \(V_{gs}\)
global variation + local variation (All MC)
local variation (Mismatch MC)
global variation (Process MC)
We had better bias mos gate with mirror rather than the vdc
source while simulating sub-block.
This is real situation due to current source are always biased by
mirror and vdc biasing don't give the right result in global
variation Monte Carlo simulation (542.8n is too pessimistic,
13.07p is right result)
For any given constant values of u and v, the
constant values of variables that solve the the feed back relationship
are called the operating points, or equilibrium
points.
Operating points can be either stable or
unstable.
An operating point is unstable if any or some small perturbation near
it causes divergence away from that operating point.
If the loop gain evaluated at an operating point is less than
one, that operating point is stable.
This is a sufficient condition
With \(m_{1\to 2} = 1\)\[
\text{Loop Gain} \simeq \frac{V_{BN}-V_{T2}}{V_{BN}-V_{T2} + V_R}
\tag{$LG_0$}
\] Assuming all MOS in strong inv operation, \(I\), \(V_{BN}\) and \(V_R\) is obtain \[\begin{align}
I &= \frac{2\beta _1 + 2\beta _2 - 4\sqrt{\beta _1 \beta
_2}}{R^2\beta _1 \beta _2} \\
V_{BN} &= V_{T2} + \frac{2}{R\beta _2}(1- \sqrt{\frac{\beta
_2}{\beta _1}}) \\
IR &= \frac{2}{R}\left( \frac{1}{\sqrt{\beta_2}}
- \frac{1}{\sqrt{\beta_1}} \right)
\end{align}\]
Substitute \(V_{BN}\) and \(V_R\) of \(LG_0\)\[\begin{align}
\text{Loop Gain} & \simeq
\frac{1-\sqrt{\frac{\beta_2}{\beta_1}}}{\frac{\beta_2}{\beta_1} -
3\sqrt{\frac{\beta_2}{\beta_1}}+2} \\
&= \frac{1}{2-\sqrt{\frac{\beta_2}{\beta_1}}} \tag{$LG_1$}
\end{align}\]
E. Alvarez-Fontecilla, P. S. Wilkins and S. C. Rose, "Understanding
High-Resolution Dynamic Element Matching DACs [Feature]," in IEEE
Circuits and Systems Magazine, vol. 23, no. 4, pp. 34-43,
Fourthquarter 2023
E. Alvarez-Fontecilla and P. S. Wilkins, "Linearity Through Democracy
[Feature]," in IEEE Circuits and Systems Magazine, vol. 25, no.
1, pp. 58-69, Firstquarter 2025
Autozeroing
offset is sampled and then subtracted from the
input
Measure the offset somehow and then subtract it from the input
signal
then \[
\Delta I_1 = \frac{1}{2}(V_{a1} - V_{b1})(g_{m,a1}+g_{m,b1})
\] That is, \(g_{m,a1}+g_{m,b1} = \mu
C_{OX}\frac{W}{L}(V_{a1}+V_{b1} - 2V_{TH})\)
To minimize the difference between \(\Delta
I_1\) and \(\Delta I_0\), the
drift of both differential and common mode between \(V_a\) and \(V_b\) shall be alleviated
Chopping
offset is modulated away from the signal band and
then filtered out
Modulate the offset away from DC and then filter it out
Good: Magically reduces offset, 1/f noise, drift
Bad: But creates switching spikes, chopper ripple
and other artifacts …
Chopping in the Frequency
Domain
Square-wave Modulation
definition of convolution \(y(t) =
x(t)*h(t)= \int_{-\infty}^{\infty} x(\tau)h(t-\tau)d\tau\)
The Fourier transform of \(s(t)=x(t)x(t)\), and we know \[\begin{align}
S(j2n\omega_0) &= \frac{1}{2\pi}\int X(j(2n\omega_0
-\omega))X(j\omega) d\omega\\
&= \frac{1}{2\pi}\int X(j(\omega-2n\omega_0))X(j\omega) d\omega
\end{align}\]
lower effective gain: DC level at the output of the
amplifiers is a bit less than what it should be
chopping artifacts at the even harmonics: frequency of
output is \(2f_{ch}\)
Below we justify \(A_\text{eff} =
A(1-4\tau/T_\text{ch})\)\[
V_o(t) = A + (V_0-A)e^{-t/\tau} \qquad V_o(T/2) = -V_0
\]
then \[
V_0 = -A\frac{1-e^{-T/2\tau}}{1+e^{-T/2\tau}}
\] Then DC level is \[
A_\text{eff} = \frac{1}{T/2}\int_0^{T/2} V_o(t)dt =
A\left(1-\frac{4\tau}{T}\cdot
\frac{1-e^{-T/2\tau}}{1+e^{-T/2\tau}}\right)\approx
A\left(1-\frac{4\tau}{T}\right)
\]
Nicola Da Dalt , Understanding Jitter and Phase Noise: 3.1.3 Voltage
to Excess Phase Transformations: Random Noise
Given \(\color{blue}\phi(t)\ll 1\),
the autocorrelation still depends on absolute time \(t\). Therefore \(v(t)\) is cyclostationary,
and they must take a time average over one carrier
period.
Chembiyan T. Jitter and Phase Noise in Phase Locked Loops [link]
\[
y(t) = A\cos(2\pi f_0t+\phi_n(t)) \approx A \cos(2\pi f_0 t) - A \phi_n
(t)\sin(2\pi f_0 t)
\]
\[
R_x(\tau) = \frac{A^2}{2}\cos(2\pi f_0\tau)
+ \frac{A^2}{2}R_\phi(\tau)\cos(2\pi f_0\tau)
\] The PSD of the signal \(x(t)\) is given by \[
S_x(f) = \mathcal{F}\{R_x(\tau)\} =
\frac{P_c}{2}\left[\delta(f+f_0)+\delta(f-f_0)+S_\phi(f+f_0)+S_\phi(f-f_0)\right]
\] where \(P_c = A^2/2\) is the
carrier power of the signal
modulated with full-cycle
random
Given \(\color{blue}\Theta\sim
U[0,2\pi]\), after ensemble averaging, the autocorrelation
becomes WSS
Haykin, Simon S., and Michael Moher. Communication Systems.
5th ed. John Wiley & Sons, 2009. - Mixing of a Random Process
with a Sinusoidal Process
Assume \(\Theta\sim \mathcal
U[0,2\pi)\) is a single random phase that remains
constant for all \(t\): \[
\color{blue}\boxed{X(t)=A\cos(\omega_0t+\Theta).}
\]
For \(t_1\) and \(t_2\), \[
R_X(t_1,t_2)
=
\mathbb E[X(t_1)X(t_2)].
\] Using \(\cos a\cos
b=\frac{1}{2}\left[\cos(a-b)+\cos(a+b)\right],\)
we obtain \[
\begin{aligned}
R_X(t_1,t_2)
&=
\frac{A^2}{2}
\mathbb E\left[
\cos\big(\omega_0(t_1-t_2)\big)
+
\cos\big(\omega_0(t_1+t_2)+2\Theta\big)
\right].
\end{aligned}
\] Since \(\Theta\) is uniform,
\[
\mathbb E\left[
\cos\big(\omega_0(t_1+t_2)+2\Theta\big)
\right]=0.
\] Therefore, \[
\boxed{
R_X(t_1,t_2)
=
\frac{A^2}{2}
\cos\big(\omega_0(t_1-t_2)\big)
}
\] or, defining \(\tau=t_1-t_2\), \[
\boxed{
R_X(\tau)=\frac{A^2}{2}\cos(\omega_0\tau)
}.
\] The autocorrelation depends only on the time difference \(\tau\).
Is it WSS?
Yes. The two WSS conditions are satisfied: \[
m_X(t)=0,
\] which is constant, and \[
R_X(t_1,t_2)=R_X(t_1-t_2).
\] Thus, \[
\boxed{X(t)\text{ is wide-sense stationary.}}
\] In fact, because a time shift simply changes the uniformly
distributed phase, \[
X(t+t_0)
=
A\cos\left(\omega_0t+\underbrace{\Theta+\omega_0t_0}_{\text{still
uniform modulo }2\pi}\right),
\] the process is also strict-sense
stationary.
Power spectral density
Using the angular-frequency Fourier-transform convention \[
S_X(\omega)
=
\int_{-\infty}^{\infty}
R_X(\tau)e^{-j\omega\tau}\,d\tau,
\] and \[
\mathcal F\{\cos(\omega_0\tau)\}
=
\pi\left[
\delta(\omega-\omega_0)+\delta(\omega+\omega_0)
\right],
\] we get \[
\boxed{
S_X(\omega)
=
\frac{\pi A^2}{2}
\left[
\delta(\omega-\omega_0)
+
\delta(\omega+\omega_0)
\right]
}.
\] Thus, the PSD consists of two spectral lines at \(\omega=\pm\omega_0\).
The total average power is \[
R_X(0)=\frac{A^2}{2},
\] and equivalently, \[
\frac{1}{2\pi}\int_{-\infty}^{\infty}S_X(\omega)\,d\omega
=
\frac{A^2}{2}.
\] For frequency \(f\), where
\(f_0=\omega_0/(2\pi)\), \[
\color{blue}\boxed{
S_X(f)
=
\frac{A^2}{4}
\left[
\delta(f-f_0)+\delta(f+f_0)
\right]
}.
\] The random phase makes the ensemble
stationary; a sinusoid with a fixed deterministic phase is not
normally regarded as a stationary random process because it contains no
random ensemble.
modulated with
deterministic cosine
the carrier phase/time origin is fixed, not randomized, it not WSS
but cyclostationary
Thus an important result is: multiplying ideal white noise by
a sinusoid does not change the shape of its time-averaged PSD—it remains
white—but its average PSD is reduced by a factor of 2,
because
\[
\overline{\cos^2(\omega_0t)}=\frac12
\]
For colored noise, however, the sinusoidal multiplication creates two
shifted copies:
Alan V Oppenheim, Ronald W. Schafer. Discrete-Time Signal Processing,
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R. E. Ziemer and W. H. Tranter, Principles of Communications, 7th
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Rhee, W. and Yu, Z., 2024. Phase-Locked Loops: System
Perspectives and Circuit Design Aspects. John Wiley & Sons
Lacaita, Andrea Leonardo, Salvatore Levantino, and Carlo Samori.
Integrated frequency synthesizers for wireless systems.
Cambridge University Press, 2007
Phillips, Joel R. and Kenneth S. Kundert. "Noise in mixers,
oscillators, samplers, and logic: an introduction to cyclostationary
noise." Proceedings of the IEEE 2000 Custom Integrated Circuits
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